Hi Steve, thanks for your response... I will try it this saturday and I'll let you know...
Best regards On Wed, Jun 11, 2008 at 7:11 AM, Steve Totaro < [EMAIL PROTECTED]> wrote: > On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <[EMAIL PROTECTED]> > wrote: > > Hi list, > > > > I'm having trouble with calls placed to the PSTN (through a TDM card), > > sometimes (a lot indeed) when I dial a number the callee party can't hear > me > > at all. > > > > My setup is: > > > > Asterisk 1.4.20.1 > > Zaptel 1.4.11 > > libpri 1.4.4 > > Wanpipe 3.2.4 > > > > I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream GXP-2000 > IP > > Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel > > 2.4.16.60-0.23-smp > > > > I'm using the ulaw audio codec. > > > > There is no NAT between the Asterisk Server and the Phones (the phone and > > the server are in the same network segment). > > > > What can it be??? > > > > Thanks in advance for any help/comment... > > > > > > -- > > Raul > > Linux Counter #156439 > > Is your Asterisk box dual homed? Firewalled? Any output from the CLI > with verbose turned on, that might help? Turn on SIP debugging as > well. > > Thanks, > Steve T > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Nacho Linux Counter #156439
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
